[linux-audio-user] lame and low sample-rate mp3's

I am writing here because my problem involves many tools. I have been trying to make extremely compact mp3's for download. $lame PO_audio.wav -m m -b 8 PO_audio-low.mp3 LAME version 3.93 MMX (http://www.mp3dev.org/) CPU features: i387, MMX (ASM used), SIMD Autoconverting from stereo to mono. Setting encoding to mono mode. Resampling: input 32 kHz output 8 kHz Using polyphase lowpass filter, transition band: 2742 Hz - 2839 Hz Encoding PO_audio_kino.wav.wav to PO_audio-low.mp3 Encoding as 8 kHz 8 kbps single-ch MPEG-2.5 Layer III (16x) qval=2 Frame | CPU time/estim | REAL time/estim | play/CPU | ETA 1038/1040 (100%)| 0:05/ 0:05| 0:05/ 0:05| 13.322x| 0:00 average: 8.0 kbps Everything looks right except that when I play the resulting file in xmms or mgp123 it plays much too fast. Could the headers on the mp3 be wrong? how do I check? they show up correctly in xmms and mpg123. mpg123 PO_audio-low.mp3 ( ... ) Playing MPEG stream from PO_audio-low.mp3 ... MPEG 2.5 layer III, 16 kbit/s, 11025 Hz mono I know I used to play lower than CD quality sample rates. Does anyone have an idea what is going on? I have tried differing combinations of ecasound and lame with combinations of -sr and -rate switches, to often similar results. I am using the alsa drives from cvs for my audiofile 2496, Thanks, --Marco